We do, thanks again!

MC, with  Digital DC Protection at default, no need to set the high pass filter anymore?

Is this the final firmware update?

> Some visualization tuning: the horizontal level meter got 10 dB markers...

ummm... am I the only person who dislike this change (imho it's not pretty because it does not scale linearly...)? can we get an option for not showing it?

it doesn't really make it easier to read  because there's no labels such as -10, -20...,
and to get the numbers, the more straightforward thing is to read the number directly on the right instead.

With DCP set to Filter, yes.

Surely matter of taste and configurable would be an option ..

But I do not fully understand the markers values/scale. According to the level meter for "L" channel the value is "-31.3", so the 4th marker from right is -30dB. It appears as if the distance between -20 and -10dB marker seems to be too high or what are the numbers behind the markers? For me it looks at the moment as if this wouldn't fit.

Yes. Maybe 0dB should be a thicker vertikal bar (2 instead of 1 pixel) or alternatively should have red color so that the marker can stay a thin and precise line.

Well, yes, I compared it at DigiCheck ...

Do you mean the B/T encoders no longer work while the unit is at a preset? Wil it work again if not at preset?

According to the latest manual v3.0:
Filter adds a special, smooth 1st order high pass to the signal path, which shows zero delay (latency), very low distortion and lower phase deviation as standard digital filters. This filter intentionally not only removes DC, but also attenuates the inaudible, but often to find infrasound junk
through a corner frequency of 7 Hz. The filter operates in all sample rates, but not in DSD mode.

I guess it is permanent, if I understands it wrong, MC will correct me.

So, suppose we merge, or load B/T with preset to make it a 7 band PEQ, then when we want to alter the B/T bands, we should do it first and then load it to the PEQ page, right?

(edit) problem/question solved by my ownself by trying/experience it.

Sorry for my English, I should write, load it to the PEQ preset particular page.

I mean we set the desired B/T value first, and then select Load B/T to preset on, and store it to a particular preset (1-10 ...) to make it a fixed PEQ graph with 7 bands.

P.S.
Ar .. after several try and experienced it, now I understand how it works, question answered by my ownself....

Always possible at the DAC in the I/O submenue:
I/O -> Main Output 1/2
        | Phones Out 3/4 -> Bass Treble -> B/T Enable -> ON | OFF

Alternatively by remap key. I am using this key remappings for the ADI-2 Pro:
Options -> SPDIF / Remap Keys -> Remap Keys ON
1) Options -> SPDIF / Remap Keys -> VOL (1)      -> Toggle View
2) Options -> SPDIF / Remap Keys -> I/O (2)       -> Toggle Ph/Line
3) Options -> SPDIF / Remap Keys -> EQ (3) -> EQ+B/T+Ld 1-4
4) Options -> SPDIF / Remap Keys -> Setup (4)   -> Mute all

to 1) useful to switch between different displays using remote
to 2) to toggle between monitors/phones when phones are permanently plugged
to 3) useful to turn off/on all potential sound manipulations (in this case to reactivate Bass why I mention this here)
to 4) if you sit next beside the ADI-2 to be able to mute quickly

Just like ning I dislike the new horizonzal level meter too...an option to switch between the old and the new one would be cool.

More details if you want help. Is the option in brackets? DSD playback or highest sample rate active?

Now with Audirvana the B / T works if I use the imac it deactivates the BT control and goes in brackets ...

I remember that manual states clearly that at high sample rates B/T & EQ can not be enabled at the same time

I think your suggestion looks perfect.

For me it would be only usable with attenuators because I'm mostly listening with about -40 db in the nearfield at the moment which means that one or two markers would be enough.
Without attenuators I would have to change the volume and the phase of the subwoofer to reach -20 or -10 db. The connected monitors are not the issue because I could adjust the variable gain.

Yes maybe because of this color. Another thing: the pre and post-fx levels are not that high in my case as I use replay gain tags to listen to music at an equal level.

After I looked at it now more often, what values the markers represents, its after some time easier for me to see/remember.

Was just an idea.

Internal distortion problem, which I was introduced with 250 / 102 and got rid of in previous 259 / 111 -version, seems to be back in this newest firmware. It happens on following conditions:

-ADI-2 Pro FS R
-more than -8.0 -volume
-analog input as source
-PH 3/4 as output
-Basic mode: DAC
-loudness: OFF
-B/T: OFF
-EQ: ON
-Crossfeed: 5
-Hi-Power
-analog input ref. level +24 dBu
-sample rate 192 kHz
-EQ´s in use:

     -B1: +6.0 dB, 25 Hz, Q 0.7
     -B2: -2,5 dB, 150 Hz, Q 0,5
     -B3: -2.0 dB, 1,33 kHz, Q 1.4
     -B4: -3.0 dB, 3.2 kHz, Q 2.2

-other settings on default

-most dynamic movie soundtrack on playback, Blade Runner 2049 opening -scene, for example

Results sharp snapping -type distortion sound (digital clipping) in loudest parts of the signal. Level meters won´t show clipping, maximum level incoming to analog input stays -1,4 dB below clip. Analog feeder used is Yamaha RX-V575 -avr from its headphone output, headphone is Sennheiser HD 600, test signal said Blade Runner 2049 -opening scene & its Dolby TrueHD soundtrack.

Distortion doesn´t happen on previous 259 / 111 -firmware, on same settings and conditions, scene passes clean without clipping. In my opinion, this indicates that something in internal processing of ADI-2 Pro, running other than 259 / 111 -firmware, is causing this. Could RME professionals take a look?

By this you are driving the analog input unbalanced to +22.6dBu (+24 -1.4).

I just measured this and in these extreme cases (balanced input driven unbalanced to max level) the input sinewave clips at +22.5dBu, so you are loosing 1.5dB of headroom from this unusal drive case.
Given the movie soundtrack will be low frequency rumbles with high peak values it is likely that you're just right on the edge of clipping the input. Hit and miss.

Firmware has no effect in this, it's a purely analog problem.

Turning the volume a bit down on the Yamaha would be the most simple fix, wouldn't it?

I tried to describe my settings... but seemingly failed. Meant to say, I haven´t done anything weird. Like bundled all EQ´s together, trying to have 5 x 12 dB (+6 dB) boost @ 20 Hz... 

Yes, phones output Hi-Power -state and over -8.0 -volume levels are very much my settings. Level meters measurement position setting is Post-FX.

+ 24 dBu on unbalanced inputs is not fully supported. This is not explicitely mentioned in the manual because usually no one has a source reaching those levels. Unbalanced is typically capped at +21 dBu peak level, everywhere. So it does make a dfference if you use balanced or unbalanced. Now your case is unusual as you use the phones output of a power amp, which is typically taken directly from the speaker output with a simple serious resistor (cheap and dirty and the reason we can sell better headphone amps...). The voltage levels available here easily overload ANY line electronics, so you DO NEED to keep the level lower at the Yamaha. To get the right level set the input to +19 dBu and watch the ADI's input level meter below the Analyser.

Regarding your other comment: You have a full scale level input signal which is then processed with EQ and easily goes above 0 dBFS. To keep this signal without distortion you are required to reduce the volume at the ADI output (Phones) so that the Phones Analyzer's Post FX level meter no longer shows overs.

If you experience differences between the two firmware versions then I suspect this is caused by an overlooked setting that behaves differently. There is definitely no difference in headroom and distortion, and especially not in analog input overdrive behaviour (firmware can't change that).

Finally: the HD600 has no sub-bass, that's why for more rumble you upped the EQ at 25 Hz. I agree. But even at modest volumes you then run in danger to hear crackling of the headphone drivers, not the driving electronics, with the sources that you mentioned.

Exactly how it is not FULLY supported? Is it PARTIALLY supported, though?? For what I know, electronic circuits care mostly about voltage levels which is put into them... So how exatcly is this RME input stage selective, if that +24 dBU comes from unbalanced or balanced feeder? My horizontal BS-detectiometer gives now almost full scale -levels...

This same Yamaha has fed, with this exactly same connection method, several other line level accepting headphone amplifiers before ADI. With zero problems. Higly doubt that input level is problem in this case either. Namely... as a recap... I haven´seen single OVR-indication from any of my ADI-2 Pro´s POST-FX -meters, input or output side, since I did level adjustments on its taking to use -installation with Yammu. Do you suppose that ADI could still, partially, hurt its analog input feelings secretly though, without showing it in its meters??

I use maximum ref. level to minimise Yamaha´s noise levels, which are not actual high end -class. +24 dBu setting gives most silent backround.
Increasing ADI´s input sensitivity starts penalty shot banging noise level -wise. For second, it doesn´t help anything to this problem. Same distortions happen, only with bigger backround noise.

What comes to my headphone EQ-settings, those very EQ-settings which were posted previously, are oratory1990´s "Optimum HiFi Curve" -settings for HD 600.

Senn HD 600, that trusty old workhorse, has in my use taken that kind of bass upshelvings for years without any other side effects but sturdy improvement to its bass reproduction. Its electroacoustics can handle that +6 dB shelf-lift @ 25 Hz with ease, just upgrading it to unbeatable newer "high-end" headphone killing machine. It doesn´t make those distortion effects for sure.

Exactly like I explained. Balanced it accepts +24 dBu, unbalanced no more than +22.5 dBu. Then the input stage will clip.

Sorry, but I have no time for electronics basics teaching lessons here in the forum. You either accept my word that it behaves like that or not.

Then I guess they all had a volume pot. Show me a headphone amp that can accept more than +21 dBu at its unbalanced input without a volume pot (that serves as input attenuator prior to any electronic circuit, thus obviously allows any input level).

As explained - on the input side yes, output I don't think so.

That was just an idea. Some people hear very loud up to the drivers clipping. I did not say that this is the guaranteed cause.

Your way of motivating us to search for an unknown issue fails miserably.

It is only a personal comment, based on my experience in dealing with documentations:

The recognition of functions/dependencies is usually much quicker and associated with fewer errors by means of a picture (here a table) than by means of a text.

A buyer (consumer, not professional) needs a device with EQ functions. Does this buyer take the effort to understand the description (without paper and pencil) or does he order the unit and try it out? I have my doubts: in Germany, free return shipping is still offered.

Just a few additions only for your information:

F7: Page 23: "Die Güte (Quality factor) ist in Band 1 bis 3 einstellbar von 0,5 bis 9,9, in Band 3 und 4 bis 5,0," change to "Die Güte (Quality factor) ist in Band 1 bis 3 einstellbar von 0,5 bis 9,9, in Band 4 und 5 bis 5,0,"

Appendix to "Peak, Shelf, High Cut and High Pass":

RME produces devices for professional use. With the ADI-2 PRO / DAC RME has now for the first time been able to convince considerably more consumers to buy them. With the ADI-2 DAC this was planned in the same way. There will be consumers who are not familiar with terms - e.g. "Peak, Shelf, High Cut and High Pass". A help could be appendices, which are inserted at the end of the document. By using graphics, the appendices would be language-neutral. The effort for referencing the attachment(s) in the text would be very low.

Extend production period for current ADI-2 PRO / DAC:

In the foreseeable future, firmware updates will only be available for special cases.
I believe that with a revised user interface, more buyers could be generated. These changes cost money and tie up resources (development, support). Only RME can find out if it could be worthwhile.

Proposals:

e.g.:
Using the rotary function of the VOLUME encoder for vertical scrolling in the "I/O | EQ | SETUP" menus: Redundant to encoder 1 + 2 (vertical scrolling) if the memory space is sufficient otherwise substitute for encoder 1 + 2. Please note: I did not study the user interface of the ADI-2 PRO.

If "E: I/O function - One idea to save storage space:" were realised, space could possibly be created for a new "Expert" function in the "SETUP" menu:
With "Expert ON", the full range of functions would be enabled. E.g. changes of parameters of outputs that are not currently activated.
...